Move to using pthread_cancel in place of pthread_kill -- due to a pro…

…blem in CYGWIN, but the move makes the code simpler and more rugged. Tidy up the code a bit and remove a clobbering warning.

audio_ao.c

@@ -121,7 +121,7 @@ static void deinit(void) {
 static void start(__attribute__((unused)) int sample_rate,
  __attribute__((unused)) int sample_format) {}
 
-static void play(void * buf, int samples) { ao_play(dev, buf, samples * 4); }
+static void play(void *buf, int samples) { ao_play(dev, buf, samples * 4); }
 
 static void stop(void) {}
 

mdns.c

@@ -106,14 +106,14 @@ void mdns_unregister(void) {
 
 void *mdns_dacp_monitor(char *dacp_id) {
  void *reply = NULL;
- if ((dacp_id!=NULL) && (*dacp_id!='\0')) {
- if ((config.mdns) && (config.mdns->mdns_dacp_monitor)) {
- reply = config.mdns->mdns_dacp_monitor(dacp_id);
- if (reply == NULL) {
- debug(1, "Error starting a DACP monitor.");
- }
- } else
- debug(3, "Can't start a DACP monitor -- none registered.");
+ if ((dacp_id != NULL) && (*dacp_id != '\0')) {
+  if ((config.mdns) && (config.mdns->mdns_dacp_monitor)) {
+  reply = config.mdns->mdns_dacp_monitor(dacp_id);
+  if (reply == NULL) {
+  debug(1, "Error starting a DACP monitor.");
+  }
+  } else
+  debug(3, "Can't start a DACP monitor -- none registered.");
  }
  return reply;
 }

player.c

@@ -545,8 +545,9 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, int64_t timestamp
  if (config.disable_resend_requests == 0) {
  rtp_request_resend(next, 1, conn);
  if ((back_step + k + resend_interval) >= sd)
- debug(2, "Last-ditch (#%d) resend request for packet %u in range %u to %u. Looking back %d packets.", j, next,
- conn->ab_read, conn->ab_write,back_step + k);
+ debug(2, "Last-ditch (#%d) resend request for packet %u in range %u to %u. "
+ "Looking back %d packets.",
+ j, next, conn->ab_read, conn->ab_write, back_step + k);
  conn->resend_requests++;
  }
  }

rtp.c

@@ -177,7 +177,7 @@ void *rtp_audio_receiver(void *arg) {
  debug(3, "Audio receiver -- Server RTP thread interrupted. terminating.");
  close(conn->audio_socket);
  */
- 
+
  debug(1, "Audio receiver thread \"normal\" exit -- this can't happen. Hah!");
  pthread_cleanup_pop(0); // don't execute anything here.
  debug(2, "Audio receiver thread exit.");
@@ -211,50 +211,50 @@ void *rtp_control_receiver(void *arg) {
  (drand48() > config.diagnostic_drop_packet_fraction)) {
 
  ssize_t plen = nread;
- if (packet[1] == 0xd4) { // sync data
- /*
- // the following stanza is for debugging only -- normally commented out.
- {
- char obf[4096];
- char *obfp = obf;
- int obfc;
- for (obfc = 0; obfc < plen; obfc++) {
- snprintf(obfp, 3, "%02X", packet[obfc]);
- obfp += 2;
- };
- *obfp = 0;
-
-
- // get raw timestamp information
- // I think that a good way to understand these timestamps is that
- // (1) the rtlt below is the timestamp of the frame that should be playing at the
- // client-time specified in the packet if there was no delay
- // and (2) that the rt below is the timestamp of the frame that should be playing
- // at the client-time specified in the packet on this device taking account of
- // the delay
- // Thus, (3) the latency can be calculated by subtracting the second from the
- // first.
- // There must be more to it -- there something missing.
-
- // In addition, it seems that if the value of the short represented by the second
- // pair of bytes in the packe is 7
- // then an extra time lag is expected to be added, presumably by
- // the AirPort Express.
- 
- // Best guess is that this delay is 11,025 frames.
-
- // uint32_t rtlt = nctohl(&packet[4]); // raw timestamp less latency
- // uint32_t rt = nctohl(&packet[16]); // raw timestamp
-
- // uint32_t fl = nctohs(&packet[2]); //
-
- // debug(1,"Sync Packet of %d bytes received: \"%s\", flags: %d, timestamps %u and
- %u,
- giving a latency of %d frames.",plen,obf,fl,rt,rtlt,rt-rtlt);
- // debug(1,"Monotonic timestamps are: %" PRId64 " and %" PRId64 "
- respectively.",monotonic_timestamp(rt, conn),monotonic_timestamp(rtlt, conn));
- }
- */
+ if (packet[1] == 0xd4) {  // sync data
+  /*
+  // the following stanza is for debugging only -- normally commented out.
+  {
+  char obf[4096];
+  char *obfp = obf;
+  int obfc;
+  for (obfc = 0; obfc < plen; obfc++) {
+  snprintf(obfp, 3, "%02X", packet[obfc]);
+  obfp += 2;
+  };
+  *obfp = 0;
+ 
+ 
+  // get raw timestamp information
+  // I think that a good way to understand these timestamps is that
+  // (1) the rtlt below is the timestamp of the frame that should be playing at the
+  // client-time specified in the packet if there was no delay
+  // and (2) that the rt below is the timestamp of the frame that should be playing
+  // at the client-time specified in the packet on this device taking account of
+  // the delay
+  // Thus, (3) the latency can be calculated by subtracting the second from the
+  // first.
+  // There must be more to it -- there something missing.
+ 
+  // In addition, it seems that if the value of the short represented by the second
+  // pair of bytes in the packe is 7
+  // then an extra time lag is expected to be added, presumably by
+  // the AirPort Express.
+  
+  // Best guess is that this delay is 11,025 frames.
+ 
+  // uint32_t rtlt = nctohl(&packet[4]); // raw timestamp less latency
+  // uint32_t rt = nctohl(&packet[16]); // raw timestamp
+ 
+  // uint32_t fl = nctohs(&packet[2]); //
+ 
+  // debug(1,"Sync Packet of %d bytes received: \"%s\", flags: %d, timestamps %u and
+  %u,
+  giving a latency of %d frames.",plen,obf,fl,rt,rtlt,rt-rtlt);
+  // debug(1,"Monotonic timestamps are: %" PRId64 " and %" PRId64 "
+  respectively.",monotonic_timestamp(rt, conn),monotonic_timestamp(rtlt, conn));
+  }
+  */
  if (conn->local_to_remote_time_difference) { // need a time packet to be interchanged
  // first...
 
@@ -455,7 +455,7 @@ void *rtp_timing_sender(void *arg) {
  request_number++;
 
  if (request_number <= 4)
- usleep(500000); //these are thread cancellation points
+ usleep(500000); // these are thread cancellation points
  else
  usleep(3000000);
  }
@@ -518,9 +518,9 @@ void *rtp_timing_receiver(void *arg) {
 
  return_time = arrival_time - conn->departure_time;
 
- uint64_t rtus = (return_time * 1000000) >> 32;
+ //uint64_t rtus = (return_time * 1000000) >> 32;
 
- if (rtus < 300000) {
+ if (((return_time * 1000000) >> 32) < 300000) {
 
  // debug(2,"Synchronisation ping return time is %f milliseconds.",(rtus*1.0)/1000);
 
@@ -683,7 +683,7 @@ void *rtp_timing_receiver(void *arg) {
  //(return_time*1000000)>>32);
  } else {
  debug(2, "Time ping turnaround time: %lld us -- it looks like a timing ping was lost.",
- rtus);
+ (return_time * 1000000) >> 32);
  }
  } else {
  debug(1, "Timing port -- Unknown RTP packet of type 0x%02X length %d.", packet[1], nread);
@@ -947,17 +947,19 @@ void rtp_request_resend(seq_t first, uint32_t count, rtsp_conn_info *conn) {
  (struct sockaddr *)&conn->rtp_client_control_socket, msgsize) == -1) {
  char em[1024];
  strerror_r(errno, em, sizeof(em));
- debug(1,
- "Error %d using send-to to an audio socket: \"%s\". Backing off for 1/16th of a second.",
+ debug(1, "Error %d using send-to to an audio socket: \"%s\". Backing off for 1/16th of a "
+  "second.",
  errno, em);
  conn->rtp_time_of_last_resend_request_error_fp = time_of_sending_fp;
  } else {
  conn->rtp_time_of_last_resend_request_error_fp = 0;
  }
 
  } else {
- debug(3, "Dropping resend request packet to simulate a bad network. Backing off for 1/16th of a "
- "second.");
+ debug(
+ 3,
+ "Dropping resend request packet to simulate a bad network. Backing off for 1/16th of a "
+ "second.");
  conn->rtp_time_of_last_resend_request_error_fp = time_of_sending_fp;
  }
  } else {